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2 просмотров56 страницClassroom notes Module I Acoustics and Electroacoustics

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Fundamentals

Alex Heldring

March 4, 2020

1

Contents

1 Introduction to Acoustics 4

1.1 The wave equation . . . . . . . . . . . . . . . . . . . . . . . . . . 4

1.2 Solutions to the wave equation . . . . . . . . . . . . . . . . . . . 6

1.2.1 Plane wave solution . . . . . . . . . . . . . . . . . . . . . 7

1.2.2 Spherical wave solution . . . . . . . . . . . . . . . . . . . 8

1.3 Characteristic Acoustic Impedance . . . . . . . . . . . . . . . . . 9

1.4 Energy and Power . . . . . . . . . . . . . . . . . . . . . . . . . . 9

2 Frequency Spectrum 11

2.1 Frequency . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 11

2.2 Octaves . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 11

3.1 Effective Sound Pressure . . . . . . . . . . . . . . . . . . . . . . . 15

3.2 Spectral density . . . . . . . . . . . . . . . . . . . . . . . . . . . . 16

3.3 Sound Pressure Level . . . . . . . . . . . . . . . . . . . . . . . . . 16

3.4 Frequency Weighting . . . . . . . . . . . . . . . . . . . . . . . . . 17

3.5 Sound Pressure Meter . . . . . . . . . . . . . . . . . . . . . . . . 21

3.6 Time averaging . . . . . . . . . . . . . . . . . . . . . . . . . . . . 21

4 Classification of Sounds 25

4.1 Noise . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 25

4.2 Spectrum of the human voice . . . . . . . . . . . . . . . . . . . . 26

4.3 Spectrograms . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 28

5.1 Directivity . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 34

5.2 Line sources . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 35

6.1 Incoherent sums . . . . . . . . . . . . . . . . . . . . . . . . . . . 37

6.2 Coherent sums . . . . . . . . . . . . . . . . . . . . . . . . . . . . 37

7.1 Attenuation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 40

7.2 Reflection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 40

7.2.1 Plane waves . . . . . . . . . . . . . . . . . . . . . . . . . . 42

7.2.2 Spherical waves . . . . . . . . . . . . . . . . . . . . . . . . 44

7.3 Diffraction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 45

7.3.1 Fresnel zones . . . . . . . . . . . . . . . . . . . . . . . . . 46

2

8 Low frequency theory: Acoustic Impedance 51

8.1 Mechanical Impedance . . . . . . . . . . . . . . . . . . . . . . . . 52

8.2 Acoustic Impedance . . . . . . . . . . . . . . . . . . . . . . . . . 53

8.3 Helmholtz resonators . . . . . . . . . . . . . . . . . . . . . . . . . 53

8.4 Acoustic Filters . . . . . . . . . . . . . . . . . . . . . . . . . . . . 55

3

1 Introduction to Acoustics

1.1 The wave equation

Acoustics is the study of the physical phenomenon that local disturbances of the

pressure in a gas, fluid or solid medium, will propagate through that medium.

In the special case that the medium is the air that surrounds us at sea level, and

that the frequency content of the disturbance lies within the range that can be

detected by the human ear, this phenomenon is called sound. In this course we

shall be focusing on the latter case, that is, audible sound in the air at sea level,

without forgetting that it is just a small part of the general field of acoustics.

Starting with a homogeneous gas in equilibrium filling an infinite volume (no

boundaries anywhere) with a spatially and temporally constant static pressure,

density and temperature (eg. air at 105 Pa or approximately 1 Atmosphere, at

20o Celcius = 293o Kelvin and a density of about 1.2 kg/m3 ), we need to setup

the equations that describe the evolution of the pressure in time and space when

a forced disturbance takes place somewhere in this gas. In order to do this we

introduce another variable, the particle velocity → −u.

Like any gas, the air consists of a huge density of particles, mainly (>99%)

nitrogen and oxygen molecules, in total about 25×1015 molecules per mm3 .

These molecules bounce around and into each other continuously with an aver-

age speed that depends on (or rather, causes) the pressure and the temperature

of the gas. The pressure p is the average force working on a surface of one

square millimeter exerted by all the molecules that hit it, in other words, p is a

macroscopic quantity. For air at sea level the average speed of the individual

particles is about 500 m/s. But the vector average of the velocities of all the

particles in a cubic millimeter is zero if the gas is in equilibrium. The particle

velocity introduced above represents this vector average, over a small region in

space, but still containing millions of particles. It has nothing to do with the

speed of the individual molecules. Just like the pressure, the particle velocity →−

u

is also a macroscopic quantity. Unlike the pressure, which is a scalar quantity,

the particle velocity is obviously vectorial.

At any point inside a gas, the mechanical state is described by the scalar

pressure field p(~r, t) and the vector particle velocity field ~u(~r, t). The time

evolution of these fields is governed by two fundamental laws of physics. The

first one is the law of conservation of momentum (Newton’s second law):

∂→

−u

−∇p = ρ . (1)

∂t

The second equation is the law of conservation of mass:

∂ρ

−ρ∇ · →

−

u = . (2)

∂t

We now have two equations relating three variables, the third one being

the density which, like the pressure, is a scalar field ρ(~r, t). To obtain a third

equation we need to know that the propagation of sound in air is very nearly an

4

adiabatic process: The only transport of energy that takes place is an exchange

of kinetic energy due to the molecules elastically bouncing against each other.

Furthermore, the perturbations of p and ρ are very small in comparison to

their static background values. Then, there is a linear relationship between the

change of pressure and the change of density at all points:

∂p p0 ∂ρ

=γ (3)

∂t ρ0 ∂t

where p0 and ρ0 are the static pressure and density respectively and γ is a

dimensionless constant for a given gas (it is called the heat capacity ratio and

its value is about 1.4 for air)1 . The factor Ks = γp0 is called the adiabatic

bulk modulus of a gas, which is the inverse of the compressibility. A further

approximation that is perfectly valid in our case is to replace the density in (1)

and (2) by its static value ρ0 . Now, if we substitute (3) in (2) and then take

the divergence of (1) and the time derivative of (2), we can combine the two

resulting equations by eliminating the particle velocity, and obtain:

∂2p

c20 ∇2 p = , (4)

∂t2

p

as we will see below, the speed of sound. Equation (4) is the Helmholtz wave

equation. If, instead of the above operations, we take the gradient of (1) and

the time derivative of (2), we find that →−

u obeys almost the same wave equation,

the only difference is that it becomes a vectorial instead of a scalar equation.

Because the wave equation only contains derivatives of the total pressure,

not the total pressure itself, in a homogeneous medium we can write the total

pressure ptot as the sum of the static background pressure p0 and the acoustic

pressure p = ptot −p0 which equals zero when the medium is in equilibrium. The

acoustic pressure is equal to the aforementioned perturbation of the pressure.

Table 1 illustrates the claim that the perturbation is indeed very small compared

to the background pressure. Also shown are some typical values of the maximum

particle velocity and the maximum particle displacement, which is again a

macroscopic quantity, representing the average distance, in the direction of → −

u,

of millions of particles, from a ficticious static equilibrium position.

The speed ofpsound, as we have seen above, depends on the type of gas, and

on the relation p0 /ρ0 . The air behaves very closely as an ideal gas, for which

the following equation of state (the ideal gas law) is valid:

p = ρRspec T (5)

in which Rspec is the specfic gas constant, whose value only depends on the

type of gas and T is the temperature in Kelvin. Eq. (5) shows that the speed of

sound in depends on the square root of the absolute temperature. In the air a

1 Eqs.(1) to (3) are called Euler’s equations of motion. They are approximations to the

Navier-Stokes equations of motion in fluids, only valid for small perturbations in an ideal gas.

5

~

|p|max (P a) |~u|max (m/s) δ (m)

max

pain threshold 200 0,5 8 × 10−5

minimum audible sound 2 × 10−5 5 × 10−8 10−10

Table 1: Limits of pressure (p), particle velocity (~u) and particle displacement

~δ amplitudes for sound waves (at the reference frequency of 1kHz , see section

2). Compare with the static pressure in air at sea level of 105 P a and the average

speed of individual molecules of 500 m/s at T = 20o C.

is given by

c0 = 331 + 0.6T [o C] m/s. (6)

In other media, the speed of sound can be very different. Without entering

into the physics of sound propagation in media other than near-ideal gases, Fig.

1 shows the speed of sound in a range of different media to give a global idea.

Note that in solid media and to a lesser degree in liquid media, the speed of

sound is generally much higher than in air.

In order to solve (4) for any specific problem, we need to supply it with sources

and boundary conditions (restrictions imposed on p by obstacles, enclosures,

discontinuities in de medium). In general we cannot derive an exact analytical

solution for an arbitrary source and arbitrary boundary conditions. However,

since (4) is a linear equation (only sums of terms containing p to the power

one), we know that any sum of functions that are a solution of (4), is again a

solution of (4). Therefore, we can try to find simple homogeneous solutions (in

an infinite homogeneous medium) with a single, simple source, and then try to

6

construct the specific solution as a sum of these simple solutions such that the

sum fulfills the boundary conditions of the actual problem.

There are two simple homogeneous solutions that serve this purpose well.

The first one is the plane wave solution, which is exact when the source is

infinitely far away from the region under study, but it is often a very good

aproximation when the source is ’sufficiently far’. The second one is the spherical

wave solution, which is exact when the source is a point-source. In many cases

the source may be approximated by a single point source or a sum of point

sources. In some situations the plane wave model is more appropriate, in others

the spherical wave model.

If the source is far from the region where the solution for the pressure field is

sought, the orientation of the disturbance of the pressure that it causes will

be aproximately the same throughout this region. If we choose a coordinate

system aligned with this orientation, for example such that the disturbance is

only along the x-axis, then the wave equation becomes one-dimensional:

In (8), f1 is any function that, if it has a given profile along the x-axis at

time t = 0, then at t = 4t it has exactly the same profile, but shifted along

the x-axis over a distance 4x = c0 4t. The disturbance has moved, unaltered,

at a speed of c0 , in the positive x-direction. Likewise, f2 moves in the negative

x-direction. f1 and f2 are called acoustic plane waves, and we now see why c0

is called the speed of sound. Of course, for a single source we will only have one

term. We can choose the x-axis such that f = f1 . Then p(x, t) = p(c0 t − x).

Typically, sound sources emit transient signals: Initially the (acoustic) pres-

sure is zero, then the source starts emitting and finally it returns to silence. So,

at every point x, p is non-zero only during a finite period. However, every

function of a single variable, finite or infinite, can be Fourier transformed to

obtain a decomposition in a finite or infinite number of components that vary

sinusoidally with that variable. In the case of a plane wave:

X ωi

p(c0 t − x) = pi sin( (c0 t − x) + ϕi ) (9)

i

c0

where the sum is over all the different f requency components present in the

time signal p, each with its own radial frequency ωi , its own amplitude pi and

its own relative phase ϕi . If the time signal is transient, the frequency decom-

position is a continuous function of ω and the sum becomes an integral. If it

is periodic, there is a finite number of discrete frequency components. This de-

composition is very important in acoustics, for several reasons. Firstly, sound is

7

often studied as a function of frequency, because the human ear is very sensitive

to the frequency content of acoustic signals. Also, the interaction of sound with

matter is very frequency dependent. Furthermore, some important classes of

sound, for example music, are dominated by a few discrete frequencies or even a

single frequency. Such signals are called harmonic waves. Lastly, when a source

emits a signal with a constant frequency content, the pressure field around it

will reach steady state: it can be described as a function of frequency instead

of time.

We can simplify (9) by introducing the wave number k = ω/c0 , obtaining,

for each frequency component

p(x, t, ω) = pω sin(ωt − kx + ϕω ) (10)

or, equivalently, in complex notation:

p(x, t, ω) = Re{pω ej(ωt−kx+ϕω ) } = Re{Pω e−jkx ejωt } (11)

where Pω = pω e jϕω

is the complex pressure amplitude of the wave component,

called a phasor. Since we know the frequency ω of every component, we can

use the time-independent complex representation of the wave

p(x, ω) = Pω e−jkx , (12)

manipulate it (for example sum it to other waves) and then use (11) to retrieve

the true time dependent pressure.

The phasor representation of a plane wave is easily extended to arbitrary

directions of propagation as:

~→−

p(→

−

r , ω) = Pω e−j k· r (13)

→

− →

−

where k = k̂k, with k̂ a unit vector in the direction of propagation. k is called

the wave vector.

When we have a single point-source (the disturbance is created in an infinitely

small region in the gas), the field is best described in spherical coordinates with

the source at the origin. Then the solution to the wave equation does not depend

on the spherical angles φ and θ, only on the radial coordinate r, the distance

from the origin. The solution for p is then

1

p(→

−r , t) = f (c0 t − r) (14)

r

which represents a wave propagating away from the source in all directions

(the ’inward’ wave that propagates towards the source is also a solution, but

of no practical interest. The factor 1/r follows from the maths, but it is also

physically necessary for conservation of energy, as we shall see later. In the

frequency domain, (14) becomes

−jkr

e

p(→

−

r , ω) = Pω . (15)

r

8

1.3 Characteristic Acoustic Impedance

As mentioned earlier, the wave equation can also be written in terms of the

particle velocity. In the case of a plane wave along the positive x-axis, the

solution

Pω −jkx

ux (x, ω) = e , uy = uz = 0, (16)

ρc0

compare (12), complies with the wave equation and with (1) and (2) as is easily

verified by taking the appropriate derivatives of (12) and (16). The relation p

/|→

−u | = ρc0 is found for plane waves propagating in arbitrary directions. For a

plane wave, the pressure and the particle velocity are in phase, and their ratio is

a medium-dependent constant . This constant is the characteristic impedance

Z0 = ρc0 of the medium, and it plays an important role in calculations of

the reflection and transmission of acoustic waves at the boundary between two

different media, among other things. For air at 1 atm and 20o C, the value of

the characteristic impedance is approximately 413 kg m−2 s−1 .

For a spherical wave, the solution for →

−

u is somewhat more complicated:

Pω e−jkr 1

ur (→

−

r , ω) = (1 + ), uφ = uθ = 0. (17)

Z0 r jkr

The extra factor appearing in (17) with respect to (15) is a near f ield effect. As

the wave propagates away from the source the ratio between p and → −

u converges

to Z0 .

If there is more than one source, the fields due to the different sources can be

summed, as noted before. However, the sum of the particle velocities is a vector

sum; if the two waves propagate in opposite directions, the vector components

have opposite sign and cancel out. So, the relation p /|→ −

u | = Z0 only holds for

f ree f ields (in an infinite homogeneous medium) due to a single source.

Although the pressure field is necessary to obtain a complete description of the

propagation of sound in a given situation, in practise one is generally more

interested in the energy contained in the field and in particular the propagation

of energy in terms of a power density, in W atts/m2 . The acoustic energy density

at any point in the region under interest is the sum of two components, the

potential energy which depends on the acoustic pressure according to

p(t)2

wpot (t) = (18)

2ρc20

and the kinetic energy:

1 →

ρ |−

2

wkin (t) = u (t)| . (19)

2

For a plane wave (and the far field of a spherical wave), using p /|→

−

u | = ρc0 = Z0 ,

we get a total energy density of

p(t)2

w(t) = . (20)

Z0 c0

9

The power density, or the amount of energy per square meter that flows

→

−

through any point, is a vector field called the acoustic intensity I which is

related to the energy density by the law of energy conservation:

→

− ∂w

∇· I =− . (21)

∂t

Being a vector, it cannot be computed from the scalar pressure alone, we also

need the particle velocity which, obviously, has the same direction:

→

−

I (t) = p(t)→

−

u (t). (22)

For a plane wave, the power density in the direction of the wave equals

− p(t)2

→

I (t) = W/m2 (23)

Z0

u is only

valid in the far field, it can be shown that (23) is true everywhere for a spherical

wave as well. Since p is proportional to 1/r for spherical waves, (23) shows

that the power density decreases with 1/r2 . This is necessary, because the total

power radiated by a point source is homogeneously distributed over a sphere of

radius r, which has a total surface of 4πr2 .

10

2 Frequency Spectrum

2.1 Frequency

In the previous section we introduced the concept of harmonic waves of a given

frequency, and we mentioned that any sound can be decomposed into harmonic

components by way of the Fourier transform. This allows to study the different

frequencies making up the sound separately. Each frequency component has

a fixed sinusiodal time dependence sin(ωt) where ω is the radial frequency, in

rad/s. At every point in space, after every time interval 4t = T = 2π/ω, the

pressure repeats itself. T is called the period of the harmonic wave. Its inverse

is the frequency of the wave, with dimension Hertz (Hz = s−1 ). For a propa-

gating wave along the x-axis, the spacial dependence at a fixed time is sin(kx).

So, moving along the x-axis, after a distance 4x = λ = 2π/k, the pressure also

repeats itself. λ is called the wave length. Since, as we saw before, ω = c0 k, we

can relate the frequency and the wave length of a harmonic wave through

Acoustic waves in the air at sea level can theoretically have any frequency

between 0Hz and some high maximum (in the order of 10GHz 2 ), provided

there exists a source that generates those frequencies. However, we shall focus

on the frequency band between 20Hz and 20.000Hz, which corresponds to the

frequencies that the human ear can detect, the audible f requency band, also

called Sound. Acoustic waves outside this band are called infra- and ultra-

sound respectively.

2.2 Octaves

With (25) we can compute the limits of the audible band in terms of the wave

length (for c0 = 345 m/s): λmin ≈ 1, 7 cm, λmax ≈ 17 m. The wide range

of frequencies and wave lengths that are included in the audible band makes

it impractical to study sound phenomena frequency by frequency on a linear

scale. Fortunately, both the sensitivity of the human ear to frequency differences

and the frequency dependence of important acoustic phenomena such as sound

absorption, are logarithmic in nature; the higher the frequency, the less they

vary as a function of frequency. We can therefore sub-divide the full range into

sub-bands with a width that grows with the frequency. The basic unit for this

sub-division is the octave. A frequency band with a width of one octave is

any band that runs from a lower frequency fl to a higher frequency fh where

2 The mean f ree path of the molecules making up the air is about 70 nm, much longer than

the effective radius of the molecules (approximately 0,15 nm). Their average speed at room

temperature is approximately 500 m/s. The minimum period T cannot be shorter than the

average time between collisions of the molecules, or Tmin ≈10−10 s. This gives an indication

(order of magnitude) of the maximum frequency in air: 10GHz.

11

fh = 2fl . The band width fh − fl of an octave is equal to the lower frequency

fl . The center-frequency of an octave is defined as

(26)

p

fc = fl fh ,

the

√ geometric √ mean of the limiting frequencies. From (26) we see that fc =

2fl = fh / 2, so the octave is fully specified by its center frequency. The

limiting frequencies and the center frequencies of a sequence of octaves follow a

geometric series with common ratio 2 (sn = 2n s0 ). Observe that the geometric

series is exponential. Therefore its logarithm grows linearly and on a logarithmic

scale, the distance between center frequencies is constant.

If a more refined sub-division is needed, steps of a fraction of an octave are

used. A common choice are third-octaves (also written as 1/3-octaves). These

subdivide each octave into three parts that have equal width on the logarithmic

scale. The centre and limiting frequencies of the third-octaves follow the same

rules as the full octaves, to the power 1/3:

p 1/3

fh = 21/3 fl , fc = fl fh = 21/6 fl . (27)

Another common choice for even more frequency resolution are 12th-octaves.

Full octaves are observed to have a fixed relative band width

BWoct 1

= √ ≈ 70%. (28)

fc 2

whereas third-octaves have a relative band width of

BW1/3oct 1

= √

6

≈ 23%. (29)

fc 2

A logarithmic scale with less resolution that is sometimes encountered is

the decade scale. This scale is based on a common ratio of 10, so the center

and limiting frequencies are a factor 10 apart. A decade does not contain an

integer number of octaves, but, because 210/3 ≈ 10, one decade contains almost

10 third-octaves.

Like any logarithmic scale, the frequency scale is not complete without a

ref erence value that tells us where to start (on a linear scale the reference is

typically zero, but on a logarithmic scale, zero becomes minus infinity). The

reference value should somehow incorporate the range of the aubible band, 20 −

20.000Hz. It is not possible to fit an integer number of octaves in this range.

So a different approach has been taken. The octaves are identified by their

center frequencies, so it would be convenient to have these center frequencies at

’simple’ values. The most important region of the audible band is the region

around 1.000Hz because of the role it plays in human speech. So, 1.000Hz

is chosen as the reference (it doesn’t matter that it isn’t at the beginning).

Then, with ten octaves centered at 31.25Hz to 16.000Hz, we approximately

cover the entire audible band. This sub division has been standardized by the

12

International Organization for Standardization (ISO), as shown in Fig. 2. Note

that it does not follow the geometric series exactly; the lowest band is centered

at 31.5Hz instead of 31.25Hz; thanks to this small deviation, the third-octave

center frequencies span exactly a decade.

13

Figure 2: Standardized octave bands.

14

3 Sound Pressure Level

3.1 Effective Sound Pressure

Up to now we have considered the exact instantaneous sound pressure p(t) at a

position in space. We have also seen that within the frequency band of interest,

its value may oscillate with a period of 50 microseconds (for 20.000Hz). The

same happens with the transported power, which is generally the quantity of

interest. If we want to measure it accurately, we need an awful lot of time-

samples. In practise, we usually deal with signals that are stationary over much

larger time scales, meaning that their frequency spectrum varies much slower

than the instantaneous pressure. To characterize such signals, we would want to

average out the rapid oscillations. We cannot average the instantaneous pres-

sure directly, because it oscillates around zero, but we can average the squared

pressure. This leads to the concept of effective sound pressure (also known as

root-mean-square or RMS pressure), defined as

v

u T

u Z

u1

pef,T = t p(t)2 dt, (30)

T

0

in the signal. Eq. (30) gives the total effective pressure of the signal. Often

we are interested in the effective pressure per frequency band (octaves, third

octaves,..). Then we first have to pass p(t) through an appropriate filter. For a

harmonic signal with complex amplitude Pω , it is easily verified that

√

pef = 1/ 2 |Pω | . (31)

For a single plane or spherical wave, the time averaged intensity or power

density through the measured point in the direction of propagation equals, ac-

cording to (23):

2

− pef

→

I = W/m2 . (32)

Z0

The time averaged power density is important, for instance if we want to know

the total power radiated by a source. If the source is stationary (the radiated

power is constant over some time interval), and it can be considered a point

source - which is always the case at a sufficiently long distance away from it, in

terms of the wave length (r/λ 1 or kr 1) -, we can measure pef at several

points around it, to cover a closed surface, preferably a sphere for computational

ease. Then we use (32) and

→

−

Z

Prad = I · dS, (33)

S

where S denotes the closed surface, to find the radiated power. If the source

is not a point source, we cannot know the direction of propagation of the wave

15

from the pressure alone. Then we also need to measure the effective particle

velocity, →

−

u ef , which is defined analogously to (30), and use the time averaged

→

−

equivalent of (22) to find I .

For general sound sources, the above procedure is the only way to determine

the radiated power.

Let us see what happens to the effective pressure of the sum of two pure tones

in steady state (T → ∞), at different frequencies:

ZT

1

p2ef,sum = lim (pω1 sin(ω1 t + ϕω1 ) + pω2 sin(ω2 t + ϕω2 ))2 dt (34)

T →∞ T

0

ZT

1

= p2ω1 sin(ω1 t)2 + p2ω2 sin(ω2 t)2 + 2pω1 pω2 sin(ω1 t) sin(ω2 t)dt. (35)

T

0

The relative phases do not affect the integral and the third term on the RHS

of (35) vanishes unless ω1 = ω2 . Another way of saying this is that different

frequency components are uncorrelated.

Eq. (35) for two different frequencies leads directly to:

The square of the effective pressure of the sum of different frequency compo-

nents is the sum of the square of the effective pressures of each component. In

other words, the total power of a signal is the sum of the power in each fre-

quency component. Eq. (36) is a version of Parseval’s teorem for two discrete

frequencies. For a continuous frequency spectrum, Parsevals teorem reads (in

terms of the frequency f instead of the radial frequency ω)

Z∞ Z∞

2 2

|p(t)| dt = |Pf (f )| df (37)

−∞ −∞

which implies that we can measure the power carried by a signal in any frequency

interval 4f and sum over all the intervals to obtain the total power. Thus, we

can characterize a signal by the power present in each octave, or third octave,

or even as a continuous spectral power density in terms of W/Hz.

The sound pressure is the fundamental quantity for the characterisation of sound

fields. However, as we have seen in Table 1, the values of practical interest

16

(globally between the weakest sound that the human ear can detect and the

sound pressure that causes unbearable pain and permanent damage to the ears),

vary by a factor of 10 million. The human ear does not perceive this variation

linearly, we never say that one sound is a million times as loud as another.

Our perception of the strength (volume, loudness) of a sound is more adjusted

to the logarithm of the sound pressure, similar to what we have seen with the

frequency. To simplify calculations, and to better reflect our perception, acoustic

engineering almost always uses this logarithmic scale. As we have seen before, a

logarithmic scale always needs a reference value. In the case of sound pressure,

this reference is chosen to be the (approximate) hearing threshold for a pure

tone of 1kHz, which equals

pref = 20µP a (38)

(this value was found through experiments on a large sample of healthy young

adult people). The Sound P ressure Level is then defined as

pef

Lp = 20 log dB SP L, (39)

pref

(adopting the convention that log without subscript always refers to log10 ).

The units of Lp are the decibel (dB). The extension SP L specifies that the

reference level (0 dB) equals 20 log(pref ). An important note: In acoustics, like

in all branches of engineering, decibels are always defined according to

P

10 log ,

Pref

where P denotes a measure of power. As we have seen, the power carried by

a sound wave is proportional to the square of the pressure. This is where the

leading factor 20 comes from in the definition of SP L in (39). Accordingly,

the intensity in a given direction through a point in a sound field can also be

expressed in decibels through

→

−

I

LI = 10 log dBI (40)

Iref

where the reference value of Iref = 10−12 W/m2 has been chosen such that for

a plane or spherical wave in air at sea level, the dB SP L value and the dBI

value are (very nearly) the same.

With Eq. (39), we can characterize a sound field at any point in space by its total

sound pressure, or its SPL. However, as mentioned before, the human ear is more

sensitive to certain frequencies in the audible band than others, so two fields with

equal SPL can in fact be perceived as very different depending on their spectral

content. We could accomodate for this by reporting the SPL as a function

17

Figure 3: Typical Sound Pressure Levels [2].

18

fc (Hz) 31,5 63 125 250 500 1k 2k 4k 8k 16k

A-weight (dB) -39,4 -26,2 -16,1 -8,6 -3,2 0 1,2 1,0 -1,1 -6,6

very similar for all humans, we can obtain a single value that captures the

perceived sound pressure level as the sum of the spectral components, assigning

a different weight to the different frequency components. On the basis of large

scale experiments, different weighting functions have been defined for different

purposes. The most important one is the so-called A − weighting, which is

almost universally used in acoustical engineering, for example in norms and

standards for noise control.

The A-weighting curve is shown in Fig. 4 together with some less important

weighting curves. The B weighting is no longer in use, the C weighting is

sometimes used for very high sound pressure levels as it turns out that the human

perception is non-linear with respect to power; at normal levels, the human is is

much more sensitive to higher frequencies, as reflected in the A weighting curve

whereas for very loud noise, this difference is much less pronounced. Sometimes

we are in fact interested in the true, unweighted level (all weights equal one).

This case is sometimes referred to as Z-weighting.

Often it is enough to measure the total sound pressure level for each octave

band or each third octave band and apply a discrete set of weights corresponding

to the average of the weighting function over the given frequency bands. Table

2 shows the discrete weights for the standard octave bands. Notice that all

weighting curves have the value one at the reference frequency of 1 kHz. As the

the human ear is most sensitive around 2 kHz, the A-weighting curve assumes

a maximum there, with a value greater than one. Below 500 Hz and above 10

kHz it drops off quickly.

In order to distinguish the weighted levels from from each other and from the

unweighted level, it is common to indicate the scale by reporting, for instance,

an A-weighted sound pressure level in dB(A). Also, the unweighted level is

usually indicated with the symbol Lp as above, while the weighted levels are

denoted with LA or LC .

Of course, the linear sum of the squared effective sound pressure in different

frequency bands does not translate to a sum of levels directly. To find the total

SPL of a signal for which (for example) the levels per octave are known, we need

to convert to linear scale, then sum the contributions, then reconvert to dB, as

in X

Lp,tot = 10 log( 10Lp,i /10 ) (41)

i

where the index i runs over all the relevant octave bands. The scale of (41) will

be dB SP L provided the levels per octave were also in dB SP L. If we need the

19

Figure 4: Different spectral weighting curves [3].

20

Figure 5: Simplified diagram of a Sound Level Meter [4].

X

LA = 10 log( 10(Lp,i +Ai )/10 ) (42)

i

We have seen in previous sections that the fundamental parameter for identi-

fying sound fields is the Sound Pressure (Level). The device used by sound

engineers to measure his parameter is the Sound Pressure Meter (SPM). SPMs

exist in many different forms, from sophisticated devices for official, legally

binding measurements to simple consumer gadgets (there are several applica-

tions that allow sound level measurements with modern smart-phones). Fig. 5

shows a generic scheme of the functionality of an SPM. A microphone converts

the instantaneous pressure p(t) into an electric signal V (t). This signal is then

either passed through a weighting network to apply A,C or Z weighting, or it is

separated into different frequency bands by filtering. Subsequently, after being

amplified, it enters a RMS detector which applies Eq. (30) to the time signal,

with a given time constant T . The resulting signal, which is now a sequence of

discrete values with a refresh rate T , is either displayed or sent to an external

storage device.

The time averaging operation to convert the instantaneous pressure into an

effective pressure as in (30) requires a choice of the time constant T . In order

to make sure that measurements are taken according to the same procedure

everywhere, the values of the time constant have been standardized. Sound

Level Meters usually offer two options, ’S’ for slow and ’F’ for fast. These are

defined in the international standard IEC 61672-1. For historical reasons they

21

include an exponential time-averaging factor:

v

u T

u Z

u1

pef,T (t) = t p2 (τ )e−(t−τ )/T dτ (43)

T

0

reading the result on an analogue needle-display). The time constant T in

the ’S’ mode equals 1 s. If more temporal resolution is necessary, because the

measured level varies at a time scale below one second, the ’F’ mode is chosen,

with a time constant of T = 125 ms.3 Apart from these two, many SPMs have

a ’peak’ mode. In this mode, the time constant is 50 µs, but typically only the

highest measured value over the full measurement time is retained. This value

is important in noise control because very high, short bursts of pressure can

be very damaging to the ears, even when they are too short to be consciously

noticed.

The result of a measurement in ’F’ or ’S’ mode is a time series of ’instan-

taneous’ sound pressure levels. However, we are often interested in an average

level on a much longer time scale. In noise control for example, it has long been

established, that (except in the case of very high ’peak’ levels), the quantity

that determines annoyance and, more importantly, long term damage to the

human hearing, is the total sound energy to which a person is exposed, i.e. the

sound pressure level (power), integrated over time. Even if the level varies over

the total duration of the measurement, the total energy will be adequately cap-

tured by the effective pressure as defined in (30). Hence, Sound Level Meters

generally offer the option, apart from the ’S’, ’F’ and ’peak’ modes to choose an

arbitrarily large time constant and measure a single average level for the corre-

sponding period. This value is defined as the equivalent level for the period T .

Combining (30) and (39), it is given by:

T

Z 2

1 p(t)

Leq,T = 10 log dt dB SP L. (44)

T p2ref

0

The value of T can vary from a few seconds, to measure the noise level due to

a continuously working machine, to 24 hours for the average noise level of city

traffic. Often a weighted equivalent level is required, for example LAeq,T , in

which case p(t) is subject to an A-weighting filter first.

Furthermore, modern professional Sound Pressure Meters have the option

’Short Leq ’, which measures a sequence of samples, typically each 125 ms, ac-

cording to (44), without exponential weighting. Storing these samples allows to

subsequently compute Leq,T for any T which is a multiple of 125 ms.

The equivalent level gives essential information on a sound field over a given

period in a single number. However, regularly we know that the sound pres-

sure level varies during the period of interest and we require some additional

3A third standard option, ’I’ for Impulse, is now obsolete.

22

information about this variation, without the need to know the actual detailed

time evolution of the signal. For this reason, professional SPMs usually generate

a number of additional standardized values, that may appear, for example, in

legal regulations for noise control. Apart from the maximum value during the

interval in ’S’ or ’F’ mode, which is often limited separately, below are a few of

the most commonly encountered ’complementary’ measures:

• Ld/n . Noise levels in public places often have different legal limits accord-

ing to the time of day. This can be incorporated by performing separate

measurements over different time frames, and comparing with separate

legal limits. Another approach often encountered is to measure the full

24 hours, and apply a correction factor, or penalty, to the result for the

more restrictive time frame. For example, at night-time, between 10PM

and 7AM, a penalty of 10 dB may be imposed. If the average 15h day-

time level is Ld and the 9h nighttime level is Ln , then, incorporating the

penalty in the total 24h equivalent level according to (44), yields

15 Ld /10 9 (Ln +10)/10

Ld/n = 10 log 10 + 10 . (45)

24 24

Common percentiles are L10 , L50 and L90 . They are defined as: the level

that is exceeded X% of the total duration of the measurement T . They

give a global idea of the variability of the level. If L10 is almost equal to

Leq,T , the noise is practically constant. If it is much higher, then there

are short intermittent bursts of noise in an otherwise quiet background.

• LN C , N oise Criteria Levels. Although A weighting yields a good mea-

sure of the pressure level adjusted to the sensitivity of the human ear,

it sums over all frequencies so it does not reveal any information on the

relative frequency content. It turns out that a pure tone or a narrow

band of noise around a specific frequency, in particular a high frequency,

can be much more distracting than homogeneous noise, even if the latter

has a higher LA . To take this into account, a mechanism has been in-

troduced, standardized by the ISO and other standardization institutes,

that does not sum the frequencial components, but it compares the entire

spectrum with a set of reference curves, as illustrated in Fig. 6. Then it

applies the following simple rule: LN C for the measurement in question

equals the value corresponding to the lowest curve that is higher than

the measurement everywhere. Noise criteria levels are typically used for

recommended noise limits in residential or public spaces. For example,

LN C in concert halls and theatres should not exceed 25 dB, in conference

halls 30 dB and in offices 35 dB.

23

Figure 6: Noise Criteria Curves according to ISO. The black curve is a mea-

sured spectrum that has a Noise Criteria Level of 45 dB (see text) [5].

24

4 Classification of Sounds

Different types of sound can be classified according to their spectral character-

istics, as shown in Fig. 7. A first sub-division is made between deterministic

sound and random noise according to whether the instantaneous pressure and

the corresponding power spectrum allow for an analytic description or not. In

Fig. 7 the first 4 wave forms (a) to (d) are deterministic, while (e) is random

noise. We discuss them in the order of the figure:

a) A pure tone. The pressure varies sinusoidally and the power spec-

trum has a single non zero peak at the corresponding frequency.

b) A complex sinusoid, a sum of a finite number of pure tones. The

first (lowest frequency) tone is called the fundamental frequency or

first harmonic. Often the higher frequencies are integer multiples

of the f undamental frequency, for example with most musical in-

struments, but also the human voice. If that is the case they are

called superior (second, third, etc.) harmonics. The figure shows

a combination of a fundamental frequency and its third harmonic.

Frequency components above the fundamental that are not an inte-

ger multiple are called overtones.

c) A general periodic wave form. If any continuous wave form repeats

itself after a given period, its spectrum will be discrete (frequency

components at finite intervals), although it can be infinite. The

lowest frequency present will be the period of the time signal. The

more rapid the change in the instantaneous pressure, the higher the

frequencies present. For a true block-function, with its infinitely fast

change of pressure, the frequency content goes to infinity.

d) A transitory (non periodic) signal will have a continuous spectrum,

i.e. a frequency density rather than discrete components. The

shorter the pulse width, the wider the frequency content.

e) Random noise. If the pressure at different time instances is uncorre-

lated, the signal is called random noise. Often, even if the pressure

as a function of time is unpredictable, the spectral content is ap-

proximately constant over time and can be measured. This is called

stationary random noise. When we measure the noise from indus-

trial machines, car engines, street noise, etcetera, we assume that

it is stationary. Examples of non-stationary noise are explosions, or

airplanes and missiles at take-off.

4.1 Noise

The spectral density of random noise can be any continuous function of the

frequency. However, certain simple functions are of particular interest because

they are good approximations of many naturally occuring noise phenomena, or

25

they can be generated by artificial sources for measuring purposes and they

are allow for simple analytical manipulation. These noise spectra are identified

with colors, a practise which started with the label white in analogy with optics

where white light has an approximately flat spectrum over the visible frequency

band.

Accordingly, white noise in acoustics has an approximately constant spectral

density over the audible band. Natural phenomena that are truly random, such

as the noise produced by a waterfall, are approximately white noise. Since in

acoustical engineering, measurements are often made on a logarithmic frequency

scale, most commonly in octaves, it is interesting to see how the power density

is distributed in terms of octaves. Observing that each octave spans twice the

band width of the previous one, and half that of the following one, we see that,

if the power density is constant in terms of W/m2 , the total power carried in

an octave goes up by a factor 2 or 3dB with each octave (see Fig. 8(a)). Since

the human perception of sound is closer to the logarithmic scale, white noise to

the human ear sounds as if the higher frequencies dominate.

In order to have a standardized noise spectrum that is better adjusted to

the human perception, and more appropriate for measurements per octave band,

pink noise has been introduced as a random noise that has an energy density

over the audible band according to (see Fig. 8(b))

dw

∼ 1/f. (46)

df

It is easily demonstrated that for pink noise, all octaves carry the same

power, as shown in Fig. 8. There are more standardized types of noise but

white and pink are the most important ones in acoustics.

Fig. 9 shows the spectrum (averaged over many people) of human speech. Al-

though these results are for English, they are quite similar for other languages.

For normal speech, although there is a lot of energy in the lower frequencies,

with a maximum below 500 Hz, the voice has frequential components over the

entire audible band (the measurements in Fig. 9 do not include the frequency

bands below 125 Hz and above 8000 Hz because their role is insignificant for

speech transmission). In fact, the lower frequency bands contain the fundamen-

tal frequency and the first few superior harmonics of the vowels. The vowels do

not transmit a lot of information for understanding, they mainly serve to draw

the attention of a listener and to identify the speaker. Everybody has a unique

timbre (combination of overtones with varying amplitude) and different accents

reveal themselves in the spectrum of the vowels. The power of the vowels is

typically some 12 dB higher than that in the consonants (see Table 3). The

spectrum of the consonants is concentrated in the range from 1000 Hz to 4000

Hz. They are much more important for speech intelligibility (we shall get back

to this in Module II). Fig. 10 shows the relative importance of the different

frequency bands for correct understanding of a spoken message, As determined

26

Figure 7: Classification of sound according to spectral content. The images

on the left show the time dependent instantaneous pressure and on the right

the corresponding power spectrum. The 5 different wave forms are explained in

section 4.

27

Figure 8: Linear (above) and Octave (below) Spectra of White Noise (a) and

Pink Noise (b).

(ms) content (Hz) SPL (dB) Intelligibility

Vowels 90 <500 0 Low

Consonants 20 1000-4000 -12 High

speech.

out the different octave bands.

4.3 Spectrograms

In practice, most sound phenomena are not steady state but transient; a source

starts emitting at some initial time, then it may or may not reach a stable state

where the frequency components remain constant, but eventually the source

stops and the sound dies out. Even so, we can choose a time-scale and deter-

mine the average level of all frequency bands that are present on this time scale.

The time scale, which is really the integration domain in (39), can be as short

as 50 µs as in the ’Peak’ measurements of a SLM, or it can be much longer.

Nevertheless, we will end up with several samples in time, each of which with

its own spectrum. This means that a complete measurement of an acoustic phe-

28

Figure 9: Average power spectrum of male and female speech (SPL at 1m) [6].

Figure 10: Relative importance of octave bands for speech intelligibility [6].

29

Figure 11: Spectrogram Representation

axis and a time axis. A full representation of a measurement on these three axes

is called a spectrogram. Fig. 11 shows an example where the three dimensions

are visualized in perspective. Often one is most interested in only two out of

the three components. Then, a projection of the spectrogram on one of the

three axial planes is enough. The names of these three planes are indicated in

the figure: The harmonic plane shows the (average) distribution of SPL over

the frequencies. The dynamic plane shows the evolution of the total SPL over

time, irrespective of the frequency content. It is the standard output of an SLM.

Lastly, the melody plane shows the evolution of the frequency content over time.

A more common way to visualize the three dimensional result is by using a

color- or grey- scale. Figs. 12 and 13 show two examples. The first one is a

measurement of a spoken phrase in English. The SPL level is represented by

the darkness of the graph at all time-frequency locations. The second one is

a recording of a short piece (6 sec) of violin play. The vertical scale is linear

frequency, the horizontal scale is time. The SPL is represented by a colorscale,

with yellow for the highest levels and black for the lowest. We clearly see the

many superior harmonics of each tone played by the violin.

30

Figure 12: Spectrogram of a spoken sentence. Darker lines have higher SPL

[7].

Figure 13: A spectrogram of a violin, with linear frequency on the vertical axis

and time on the horizontal axis. The coloring represents SPL [8].

31

Figure 14: Typical radiated sound powers (W) and sound power levels (dB

PWL) [9].

Although it is generally difficult to measure, with any source that produces an

approximately constant sound field over a given time period, we can associate a

given total radiated power. In Fig. 14 some typical radiated sound powers are

shown. Since, just like with typical sound pressure values, the range covering

sources often found in practice is very stretched, radiated sound power is usually

given on a logarithmic scale:

Prad

Lw = 10 log dB (P W L), (47)

Pref

where the reference value for the PWL (PoWerLevel) scale equals Pref =

10−12 W . The third column in Fig. 14 gives the PWL values corresponding

to the linear (Watt) values in the second column.

In section 1.4 we derived the formula (33) that gives the total radiated power

→

−

by a source from the intensity I on a closed surface surrounding the source. For

a point source (also called an omnidirectional source because it radiates equally

in all directions), this formula simplifies to the integration of a constant value

over a sphere of radius r:

Taking the logarithm of (48) and inserting the numerical values for Pref , pref

and Z0 for air under the usual typical circumstances, we obtain the following

useful relation between source power level and sound pressure level at a distance

r, for an omnidirectional source in free space:

32

Observe that according to (49), as we move away from the source, every time

we double the distance, we lose 6 dB of SPL.

The -11 dB term in (49) is due to the factor 10 log(4π) ≈ 11 from (48). This

is because the radiated power is distributed over a full sphere, which spans a

solid angle of 4π steradians. A source that is located very near an infinite,

flat and completely stiff surface (often a good aproximation of a loudspeaker

mounted very near a large wall), radiates all of its power into the half -space in

front of it. A half-space only spans a solid angle of 2π steradians. Thus, the -11

dB term now becomes a factor two or 3 dB smaller. Hence, formula (49) for a

source very near a large stiff surface becomes

SP L = P W L − 20 log r − 8 dB (half-space) (50)

We see that a loudspeaker radiating the same power but located near a wall will

produce 3 dB more SPL than if it were located in free space. This is simply

a consequence of the radiated power being forced to propagate into a smaller

portion of space than before. It is independent of the frequency content of the

signal. However, there is a second phenomenon associated with a source near

a wall. M irror Image theory tells us that the combination of a point source

and a rigid wall, can be modelled by replacing the wall with an image-source,

located at the same distance from the wall, on the other side, such that the

line connecting the two sources is perpendicular to the wall. This image source

radiates exactly the same signal as the original source, and the combination of

the two sources now radiates into free space. Now, if the distance between the

two sources is much smaller than the wave length, then wherever we measure

the total pressure, it will be twice as large as the pressure from the original

source alone. Hence, the radiated power will be four times larger. Of course,

since the wall is no longer there to confine the power to a half-space, half of the

radiated power is lost into the ’imaginary’ half-space behind the wall. But, we

still have twice more PWL than for the same source in free space. We see that

for low frequencies, we gain not 3 but 6 dB of SPL by placing the source near

a stiff wall. The gain in power as a function of the distance from the wall in

terms of frequency is shown in Fig. 15.

It seems curious that the interaction of a wall with a source can help the

source to radiate more power than in free space. The explanation is that the

radiation resistance seen by the source (more on this in module V on loud-

speakers) changes due to the presence of the wall. Likewise, two sources, located

closely together and radiating identical signals, will radiate four times as much

power in the low frequencies as the power radiated by either source on its own

(for example two loudspeakers side by side, driven by the same amplifier).

If a source is positioned near the intersection of two perpendicular reflecting

walls (on the floor against a wall, then the space into which it radiates spans

only π steradians:

SP L = P W L − 20 log r − 5 dB (edge), (51)

and for low frequencies, mirror image theory provides an additional 6 dB of

PWL with respect to free space. This is because the first reflecting surface is

33

Figure 15: Radiated power from a point source at a distance h from a reflecting

plane, relative to the radiated power in free space, as a function of the wave

number k [10].

replaced by one image source and the second reflecting surface is replaced by

two images, one for the original source and one for the image in the first surface.

Altogether, the equivalent in free space are four sources.

If a source is positioned near a corner where three surfaces intersect, we can

add yet another 3 dB to the SPL:

SP L = P W L − 20 log r − 2 dB (corner) (52)

and for low frequencies we have eight image sources. The SPL at low frequency

at a distance r is then 18 dB or a factor 64 higher than with the same source

in free space!

5.1 Directivity

When a source is not an omnidirectional (point-) source, the radiated field will

still be spherical at sufficient distance away from it (the f ar f ield). However

the power density will not necessarily be equal in all directions. The radiated

power may be concentrated in a single direction or in one particular plane. The

source will have a directivity pattern

I(θ, φ)

Q(θ, φ) = (53)

I0

where I0 is the average intensity. Clearly, for an omnidirectional source, Q(θ, φ)

equals one everywhere. Often one is not so much interested in the details on the

34

Figure 16: Relative Directivity of the human voice at different frequencies,

horizontal plane

directivity pattern, but only in the value at the maximum. When a source is said

to have a directivity, or a directivity f actor, of Q = 2, then this is assumed to

mean the maximum of (53). The directivity index is the logarithmic equivalent

of (53), defined as

Often, the directivity pattern is represented as a relative pattern, normalized

to the maximum. Also it is usually presented as one or more polar plots, showing

two dimensional cross-sections of the full 3D pattern.

As a general rule, sources can only be directive if their dimensions are large

compared to the wave length. Hence, the directivity is a function of the fre-

quency. Fig. 16 shows the (approximate) directivity of human speech in different

frequency bands. The source (the mouth) is small at most wave lengths in the

audible band, so the human voice is not very directive. The size of the human

head also plays a role. At the highest frequencies it is several wave lengths large

so it directs the sound propagation towards the frontal hemisphere.

Sometimes rather than aproximating a source with a point source at a given

position in 3D space, it makes more sense to approximate it with an infinitely

thin but infinitely long straight line generating the same sound everywhere along

its axis. This is equivalent to reducing the situation to a 2-dimensional model,

since the sound field is identical in every plane perpendicular to the line source,

so we only need to study one such plane. Practical examples of situations that

allow for a line source approach are motorways, busy streets or long trains.

35

Line sources are studied in cylindrical coordinates with the source at the

origin. The total radiated power is now expressed in W/m, i.e. the amount of

Watts radiated per meter along the length of the source. Rather than being

distributed over a sphere, it is distributed over a circle, so the intensity drops

not as 1/r2 but as 1/r. The intensity is the radiated power per unit length over

the circumference of the circle, so (if the line source is omnidirectional),

Prad

Ilinesource = W/m2 , (55)

2πr

which gives, in logarithmic scale

36

6 Summation of Sound Fields

6.1 Incoherent sums

We have seen in section 3.2 that two pure tones of different frequency are un-

correlated or incoherent which means that the cross-term of the squared sum

of the pressures vanishes, such that the squared effective pressure of the sum is

the sum of the squared effective pressures. By extension this works for the sum

of any number of frequency components.

Two different sound waves with given spectral densities can also be inco-

herent, even if their spectra show overlap. In fact, practically all naturally

occurring sound phenomena are incoherent if they come from different sources,

the exception being natural sources producing a pure tone such as a tuning

fork. Hence, in the general case, if at some point in space, one source produces

a sound field pef 1 and another source produces pef 2 , then the total effective

pressure will follow from q

pef = p2ef 1 + p2ef 2 . (57)

In terms of Sound Pressure Levels this translates to

Ltot = 10 log 10L1 /10 + 10L2 /10 dB SP L, (58)

N

!

X

Ltot = 10 log 10Li /10

dB SP L. (59)

i=1

Rearranging the equations, we can also subtract the contributions from dif-

ferent sources. This is often used to determine the level Ls due to a given source,

if we can only measure the level in the presence of some background noise. First

we measure the level LN of the background noise in the absence of the source

(eg. a noisy air-conditioner), and subsequently the total level Ltot . The level

due to the source is then estimated with

In practise, coherent sound fields are only encountered in the case of two sources

producing the same pure tone, or when the two sources, although spatially sep-

arated, actually produce the same signal, for example two loudspeakers driven

by the same amplifier, or one source and its reflection in a wall. When such

a situation occurs, it makes sense to study the total pressure frequency by fre-

quency. At every point in space, the total complex pressure is then the sum of

the phasors of the two fields, so the squared effective pressure equals (see Eq.

(31)):

1 2 1 2

p2ef = |Pω,tot | = |Pω,1 | + |Pω,2 | ej∆ϕ (61)

2 2

37

where ∆ϕ is the position-dependent phase difference between the two phasors.

Observe that, depending on the position, the effective pressure assumes a value

between the sum and the difference of the two separate effective values. An

interference pattern is built up throughout space with minima and maxima. In

the special case that the two fields are of equal strength, (61) reduces to

1 2

p2ef = Pω,1 (1 + ej∆ϕ ) (62)

2

with minima of zero pressure when ∆ϕ equals π, 3π, .. and maxima of 2pef,1

when ∆ϕ equals 0, 2π, ..

If the two sources are point sources emitting identical signals, then the cor-

responding phasors are given by (15) with identical complex amplitudes p̂ω and

their sum is −jkr1

e−jkr2

e

Pω,tot = Pω + . (63)

r1 r2

where r1,2 are the respective distances from the sources to the observation point.

If these distances are not too different, we can aproximate (63) by setting them

equal to an average distance r in the denominators of (63) and obtain

Pω

Pω,tot = (1 + e−jk∆r ) (64)

r

where ∆r = r2 − r1 is the path-length dif f erence of the two sources to the

observation point. We observe that the field is that of a single source, modulated

by the factor (1+e−jk∆r ) which depends on k∆r = 2π∆r/λ. Over the frequency

spectrum, this factor oscillates between the value two when ∆r = 0, λ, 2λ, ...

and zero when ∆r = λ/2, 3λ/2, ... The interference of the two sources acts as a

periodic filter. When the difference is a multiple of λ the two fields are in phase.

They interfere constructively. When the difference is a multiple of λ plus λ/2,

the interference is destructive and the two fields cancel eachother.

Figure 17 shows a typical situation giving rise to this effect. If the wall is

hard (no absorption) the amplitudes of the two signals will be practically equal

but some frequencies are filtered out while others are amplified, which leads to

a coloring of the signal. Fig. 18 shows the effect of the filter on a spectrally

flat signal, for some path-length differences. The linear scale representation

has inspired the name of this effect: the comb-f ilter ef f ect. Notice that the

maxima are at 6 dB. The pressure doubles (with repect to the single source),

so the intensity quadruples.

38

Figure 17: Interference between direct and reflected signal causes the comb

filter effect (see text).

Figure 18: Comb filter response due to interference between coherent sources

with different path-length differences (delays)

39

7 Attenuation, Reflection and Diffraction

Up to here we have exclusively treated sources in a homogeneous, lossless, infi-

nite medium (except for the reflection in a wall in the previous section). This

is often a good approximation leading to useful practical results in acoustic en-

gineering. However, equally often, we must deal with situations that cannot be

captured within this simplified model. In this section we discuss three different

phenomena that govern the behaviour of acoustic fields under more realistic cir-

cumstances; attenuation of waves in the medium due to absorption, ref lection

on walls and dif f raction of sound waves by solid obstacles.

7.1 Attenuation

In reality, no medium, including the air, is completely lossless. Some acoustic

energy gets converted into heat by complicated mechanisms at the molecular

level. For air, these losses are small so they can often be completely neglected,

leading to the lossless wave equation we’ve already seen. If they are not, they

can be incorporated in the free field solutions to the wave equation for the

intensity (plane wave, spherical wave), through an attenuation factor e−mx ,

where the attenuation coefficient m, in m−1 , is the sum of a number of rather

complicated energy absorption mechanisms, involving the molecular structure

of the medium, and its state (gas, liquid, solid). Even restricting ourselves to

air at sea level as usual, the value of m is highly frequency dependent and also

depends on the temperature and the relative humidity of the medium. Fig. 19

shows the dependence of m on the frequency and the relative humidity for a

temperature of 20ºC.

To give an idea of the consequence of air absorption in practical circum-

stances, if m = 0.0025m−1 , then the intensity of a wave decreases by about

5% or 0.2dB every 10m (remember the intensity is porportional to the pressure

squared). This may become an important factor outdoors or in very large en-

closures, like a concert hall, but in smaller enclosures it is generally negligible

compared with absorption by the walls. According to the figure, this is the

approximate value of m for 2kHz with a relative humidity above 30%4 . As a

rule of thumb, air absorption only becomes relevant above 2kHz and only in

large spaces.

7.2 Reflection

When a sound wave encounters a sudden change of medium, for example a

solid wall, or the ground beneath our feet, part of the power it carries may

penetrate into the wall, while the rest is reflected back into the space where

the wave originated. How much power is transmitted into the new medium

and how it will continue to propagate in there depends on the characteristics

of that medium, and also on the direction of the incident wave with respect to

the boundary between the two media. While this is of course a very important

4 In Barcelona, the yearly average relative humidity is 70% and it rarely drops below 65%

40

Figure 19: Attenuation coefficient for sound waves in air, sea level, 20ºC [11].

41

question in acoustics, we shall continue to restrict ourselves to the propagation

of sound in air, so our interest is mainly in what happens to the part of the

power that is reflected.

Although the word ref lection refers to the sound field re-emitted into the space

of origin by any boundary of any shape in general, when we use the word,

we usually mean specular reflection, which is reflection of plane waves by a

perfectly flat, infinite boundary between media. This seems very restrictive,

but in practice we can often assume that a general boundary is both flat and

without edges locally. We can then apply the rules of specular reflection to the

regions where these conditions hold.

The expressions for the transmitted and reflected waves, given an incident

wave, follow from the boundary conditions at the boundary between the media.

The acoustic boundary conditions are that both the pressure and the normal

component of the particle velocity are continuous across the boundary. Let us

consider the simplest case of a totally hard boundary, that does not move and

does not admit any transmitted wave. Since the wall does not move, there is

no condition on the pressure. The only boundary condition is that the normal

particle velocity has to vanish on the boundary.

Recall that a harmonic plane wave travelling in the direction kˆi is written as

~ →

− →

− Pω −j k~i ·→

−

pi = Pω e−j ki · r , u i = kˆi e r

(65)

Z0

We assume that the wall is in the plane z = 0 and the plane wave travels in the

positive z-direction. We can always choose the coordinate system such that the

wave travels in the zx-plane and we can set the phasor to Pω = 1, for simplicity:

kx

→

− 1 −j(kx x+kz z)

pi = e −j(kx x+kz z)

, ui = 0 e (66)

kZ0

kz

where kx2 + kz2 = k 2 . Now, a reasonable guess is that the reflected wave will also

be a plane wave, with the same frequency but not necessarily the same direction.

It should also have the same amplitude, because all the power is reflected back

into the medium of incidence:

mx

1 −j(mx x+my y+mz z)

pr = e−j(mx x+my y+mz z) , → −

u r = my e . (67)

kZ0

mz

sum of incident and reflected field) yields:

42

Figure 20: Snell’s law for plane waves incident on a flat wall: θr = θi [12].

mz = −kz . This corresponds to a reflected wave whose direction of propagation

makes the same angle with the normal vector n̂ of the wall as the incident wave,

and is in the plane defined by k~i and n̂ (see Fig. 20). This result is called Snell’s

law.

Generalizing to a plane wave according to (65), that is incident on a flat wall

with unit normal vector n̂, the reflected wave will obey

~ →

− →

− Pω −j k~r ·→

−

pr = RPω e−j kr · r , u i = Rkˆr e r

(69)

Z0

k~r = k~i − 2(k~i ·n̂)n̂. (70)

The complex ref lection coef f icient R is a frequency dependent parameter of

the material and the structure of the wall, as well as of the medium of the

incident wave and the angle of incidence.

If the wall is simply the boundary between two infinite media with charac-

teristic impedances Z1 and Z2 , then the reflection coefficient becomes

Z2 − Z1

R= (71)

Z2 + Z1

If the medium of the incident wave is air, then Z1 ≈ 413kgm−2 s−1 , as we have

seen in section 1.3. For a totally stiff or hard wall, Z2 = ∞ so R = 1. Conversely,

a completely soft wall has Z2 = 0 and thus R = −1. The third important case

is when Z2 = Z1 , then R = 0 and we have a perfect absorber5 . However, in

5 These values are idealized values never encountered in reality, However, any hard, heavy

solid material will approximate a totally hard wall. An approximately soft wall is harder to

find, because the characteristic impedance should be small relative to that of air, which is

already very small. The closest approximation to a perfect absorber can be found in sound

laboratories called anechoic chambers.

43

general, the wall is not an infinite medium but, for example, a brick wall, a

glass window, or a separation built up of multiple layers of material. Then, R

becomes complex (and hence frequency dependent), because of the interference

of multiple reflections inside the finite thickness layers.

As usual, we are often interested in the power carried by the reflected wave

rather than its exact pressure field. The power density of the plane wave equals

2

Ii = |pi | /Z0 , hence with (69) the power density of the reflected wave is

2

Ir = |R| Ii . (72)

The power that is ’lost’ (that does not return to the room where the incident

wave originates, irrespective of what does happen to it) is then equal to αIi

where

2

α = 1 − |R| (73)

is called the absorption coefficient of the wall.

Let’s examine the simple case of a plane wave in the air inciding perpen-

dicularly on a hard wall (R = 1). Then, the reflected wave will propagate

perpendicularly back into the air, due to (70), and its amplitude will be equal

to that of the incident wave. At the boundary, the pressure of the incident wave

and the reflected wave will be in phase and sum to twice the incident pressure.

The particle velocity of the reflected wave will be equal but opposite in sign to

that of the incident wave (because k~r = −k~i ). Hence the net particle velocity

will be zero, as it should be according to the boundary condition. As we move

away from the wall, the pressures will be more and more out of phase, until at

a distance d = λ/4, the phase difference will be π radians and the pressures of

the incident and reflected wave cancel each other out completely. At a distance

d = λ/2, the pressures are in phase again and add up to twice the original

pressure. This alternating of maxima (peaks) and zeros (nodes) continues peri-

odically from here on. Such a pattern is called a standing wave. The particle

velocity shows the same pattern, but with a zero on the wall and a maximum

at d = λ/4. If |R| < 1, a similar pattern appears, but the peaks are lower and

the nodes are not zero. Fig. 21 illustrates this case.

If the incident wave is not plane but spherical, then the reflected wave will

obviously be much more complicated. However, the total field as a point in space

due to the incident and the reflected wave is rather easy to determine thanks

to a method that we have already mentioned in section 5, the mirror image

method: Let us assume that the wall is in the plane z = 0. The space above the

wall is homogeneously filled with air, and the incident field comes from a point

source at →−

r s = (xs , ys , zs ) above the wall. We wish to find the total field at a

receiver position →−

r r = (xr , yr , zr ). According to the mirror image method we

can replace the wall by an imaginary source at → −r 0 = (xs , ys , −zs ), or in general

at the point that is the perpendicular reflection of → −

r in the plane of the wall.

The field in the space above the wall is then the sum of the two sources radiating

44

Figure 21: Standing wave pattern due to a plane wave inciding perpendicularly

on a wall with a positive reflection coefficient smaller than one [13].

into infinite free space. If the wall has a reflection coeffcient unequal to one, we

multiply the field from the imaginary source with the reflection coefficient R of

the wall. The procedure is illustrated in Fig. 22

Observe that the mirror image method automatically complies with Snell’s

law, in the sense that, if we approximate the spherical waves both from the true

source and the imaginary source as locally plane around the point where the

incident wave hits the wall, then by geometrical construction the wave vectors

k~i and k~r comply with (70), that is, the angles of the incident and reflected

propagation directions with the surface normal vector are the same.

7.3 Diffraction

Dif f raction is the perturbation of the sound field in the medium in which

it propagates when it encounters an obstacle that is not flat and/or not of

infinite extent in all directions. Two other words for the same phenomenon

are scattering and dif f usion, although they usually refer to slightly different

situations: diffraction is typically used for the sound field penetrating the region

which is blocked from the direct view of the source (the ’shadow zone’) due to

the waves ’bending’ around the edges of the obstacle. Scattering is used to refer

to the perturbation of the sound field by the obstacle in all directions. Diffusion

is used when the scattering is so dominant and unstructured that the perturbed

field no longer has any dominant direction of propagation or net power transport

through any point in space.

An exact solution for the diffraction from an obstacle only exists if the obsta-

cle is a perfect sphere made of a homogeneous material. Good approximations

exist for simple geometries like arbitrarily shaped apertures in a flat hard wall

45

Figure 22: Illustration of mirror image method in two dimensions [9].

a flat hard obstacle in free space (by a theorem called Babinet’s principle, the so-

lution to the one problem can easily be converted into the solution to the other).

The diffracted field from a general shape and (combinations of) materials can

only be calculated numerically by computer simulations.

Diffraction is a consequence of the interference of the sound field scattered

from infinitesimallly small parts of the surface of the obstacles. It is therefore

highly frequency dependent. If an object is small compared to the wave length,

diffraction will be minimal and the incident field propagates around it practically

unaltered. See Fig. 23. As a consequence, an obstacle may be large enough

to shield off high frequency components but at the same time be practically

transparent to low frequencies.

A qualitative model developed by Fresnel in the 19th century (originally for

optical waves), still often used for estimating the effects of diffraction, subdivides

the space inbetween the source and the receiving point into Fresnel-zones. They

are often defined on a plane perpendicular to the line joining source and receiver,

as in Fig. 24, although they can be defined on any surface separating the source

and the receiver. First, the length of the straight path between the source and

the receiver is determined (d = p + q in the figure). Then, for every point in the

plane, the total path length from source to receiver passing through this point

is found and the difference with the direct line is determined (4 = 4p + 4q

in the figure). Now, the first Fresnel zone is the set of points with 4 < λ/2,

the second Fresnel zone has λ/2 < 4 < λ, etc. In the perpendicular plane, this

leads to concentric circles defining the separation of zones. In the figure, the

46

Figure 23: Diffraction by cylindrical obstacles that are much smaller than the

wave length and approximately of the size of one wave length.

47

Figure 24: Fresnel zones [14].

first zone is the white disk in the centre, the second zone is the smallest black

ring, etc.

With this rule, different propagation paths through the same zone are never

more than π/2 radians out of phase with the ’central’ path (for the first zone

this is not the direct line but the paths through the points half-way between

the centre and the zone-edge), while propagation paths through adjacent zones

are on average π rad out of phase. So, on average, contributions from the same

zone interfere constructively (amplify eachother) while contributions one zone

apart tend to cancel eachother out. Fresnel constructed a screen covering the

even zones and showed that for light of the appropriate frequency, this screen

acted as a lens: a light beam from the source is focalized towards the receiving

point. The same can be done with sound.

In practise, Fresnel zone theory is mainly used to obtain a quick assessment

of the effect of an obstacle: If the objective is to create a shadow zone, then

a good ’rule of thumb’ is to ensure that a least the entire first Fresnel zone is

blocked. If on the other hand we want the effect of an obstacle to be minimized,

this theory learns us that it is not enough to ensure a ’Line of Sight’ from source

to receiver, but we should try to keep open the entire first Fresnel zone.

When (referring to Fig. 24), the distances p and q are large with respect to

the size of the obstacle (and therefore to the relevant Fresnel zone radius) the

approximate formula s

nλpq

rn = (74)

p+q

can be used to find the (outer) radius of the n-th Fresnel zone.

Although the Fresnel zone theory is very approximate, it is nevertheless

very useful because especially for relatively simple shaped obstacles (circular,

48

square) the attenuation due to blockage and the minimum path length difference

divided by the wave length are very much proportional in practical situations.

An obstacle can be assigned a Fresnel number

24

N= (75)

λ

(not necessarily an integer number), where 4 is the ’typical’ shortest path length

difference with the Line of Sight. For a perfectly circular obstacle, if it com-

pletely blocks the first Fresnel zone, then its Fresnel number equals N = 1.

As an example, we consider the very practical problem of finding the correct

height of a noise barrier between a motorway and a residential neighbourhood.

Considering the motoway as an infinite line source, and assuming that the noise

barrier will be much longer than its height, the problem is reduced to a two

dimensional problem. The geometry of the problem is sketched in Fig. 25. Fig.

26 shows the theoretical attenuation in dB as a function of the Fresnel number.

The graph is calculated by a somewhat more elaborate theory than the one

presented above, but based on the same concept of Fresnel zone diffraction.

It is used by the United States Federal Highway Administration to establish

noise barrier heights. A generally accepted rule for noise barriers is that the

attenuation should be at least -10dB at the lowest frequency with a significant

contribution to the noise. Observe that this is approximately achieved when

N = 1, i.e. the first Fresnel zone is entirely blocked.

49

Figure 25: Fresnel zone theory applied to Noise Barrier [15].

50

Figure 26: Noise barrier attenuation as a function of Fresnel number [16].

We have seen that the local ratio between the pressure (a driving force) and

the particle velocity (the resulting displacement) in a free field gives rise to a

parameter characterizing the medium, the characteristic impendance (section

1.3). If the pressure and particle velocity are approximately constant in space

over a finite surface, we can calculate the ratio of the total force acting on that

surface and the resulting volume velocity U in m3 /s. The volume velocity is

defined as the particle velocity u in the direction perpendicular to the surface,

times the surface area S. From this ratio we can define the concept of Lumped

Acoustic Impedance, or simply acoustic impedance as a characteristic of such

a surface. Although this definition does not explicitly involve the frequency,

in practice the concept is typically applicable only for surfaces that are small

compared to the wave length, hence the name low frequency theory.

A related restriction is that a surface of constant pressure an particle ve-

locity is essentially only found in a plane wave, perpendicular to the direction

of propagation. Therefore the theory only applies to (approximately) normal

incidence of (approximately) plane waves. The two conditions mentioned above

are typically found in guided wave geometries, such as acoustic ducts (tubes),

or in transducers (microphones and loudspeakers). In both cases the theory is

an enormous aid in the design and analysis of the various elements.

By definition, impedance is a complex value, that only makes sense in the

frequency domain. The real part of the impedance is called the resistance, and

it determines the amount of power that is converted into a different form of

energy, usually thermal energy. From the acoustic view point, this power is lost

or dissipated. The imaginary part is called the reactance and it determines the

phase difference between the driving force and the resulting displacement. If

51

the reactance is positive it is called inductive, if it is negative, capacitive.

Acoustic impedance is very closely related to mechanical impedance, in fact

the two are practically interchangeable, and in order to introduce the acoustic

impedance, it makes sense to start with the mechanical equivalent (there is a

similar relation with electrical impedance which we’ll mention too in case the

reader is more familiar with that).

Mechanical impedance Zm is the reaction of a solid body to a force, in terms

of the ratio of the force F and the resulting velocity of the body v

F

Zm = . (76)

v

In its simplest form the force is just point-force acting upon a point of the struc-

ture. Part of the force will result in an acceleration of the body, and part will

result in a deformation of the body (stretching or compression). Another part

is transformed into heat through internal friction and friction with the environ-

ment. The different proportions are determined by the physical characteristics

of the body. The acceleration depends on the mass M of the body through

dv

F = Ma = M = jωvM, (77)

dt

so the inductive reactance due to the mass equals

Xm = jωM. (78)

theory. The deformation depends on the stiffness (or spring constant) K of the

body through Hooke’s law:

Z

K

F = Kδ = K vdt = v (79)

jω

where δ is the change of length of the body in the direction of the force. Con-

sequently the capacitive reactance of the body equals

K

Xm = , (80)

jω

analogous to the electrical capacitance of an electrical condenser or capacitor.

These two characteristics of mechanical impedance of a solid body can be

represented as a mass hanging from a spring, as in Fig. 27. The total mechanical

impedance Zm is the sum of the two components above plus a resistance term

Rm that represents the losses though friction:

F K

Zm = = jωM + + Rm . (81)

v jω

52

Figure 27: A mass M hanging from a spring with stiffness K.

The acoustic impedance of a surface S is defined as

p

Za = , (82)

U

the ratio between the pressure and the volume velocity (both assumed to be

constant over the surface). Notice that over the surface S, the total force by

the pressure equals F = Sp. At the same time, the volume velocity equals

U = Su. We can thus consider the air just in front of the surface as a single

homogeneous body with a constant velocity equal to v = u = U/S. So, if a flat

solid object, such as the diaphragm of a loudspeaker or a microphone with a

surface S vibrates in the air, it is loaded with a mechanical impedance equal to

Zm = Za S 2 . (83)

This shows that the analogy between mechanical and acoustical impedance is

not just an artefact. It can be used to derive a formulation for the total me-

chanical impedance of a microphone or loudspeaker, that includes the radiation

impedance of the sound that is dectected or produced respectively (Ultimately,

microphones and loudspeakers convert between electric and acoustic signals.

The analogy can be extended to electrical circuits as we have seen above; then

the entire system can be described in terms of either mechanical forces and ve-

locities or voltages and currents. We shall revisit this subject in Modules IV

and V).

The acoustic impedance can also be used to model acoustic ducts or tubes

guiding acoustic signals. Practical applications are air or heat exchangers in

buildings and vehicles, or mufflers on car exhaust pipes or guns, that must all

allow a maximum air flow, while attenuating the propagation of sound to a

maximum.

A special case are acoustic wave guide sections that are very short in comparison

with the wavelength. Consider a short very thin tube of cross-section S that

53

is open at both ends. If a low frequency wave incides on one end, since the

other end is open, the air column inside it will move as a whole rather than be

compressed. It therefore represents a mechanical reactance of

(the mass is the volume times the density. The volume is of course Sl, where l

is the length of the tube, but we use an effective length lef ,slichtly longer than

l, because the moving air column extends a bit outside of the tube). Hence,

with (84), the acoustical reactance gives

Now consider a hard cavity that is small with respect to the wave length,

with an opening of cross-section S. When a low frequency incident wave incides

upon the opening, the cavity is too small to allow wave propagation, so the air

inside the cavity will be uniformly compressed. The pressure p, and density ρ

in the cavity are related through equation (3):

ρ0

ρ= p. (86)

Ks

Where Ks is the adiabatic bulk modulus from section 1.1. The volume velocity

can be related to the density through continuity of mass: Considering that the

total mass of the air in the cavity M = V ρ, and that, the cavity having hard

walls, its volume V is a constant, we have (again for small relative changes of

ρ),

∂M

= jωρV = ρ0 U. (87)

∂t

Combining (86) and (87) to eliminate the density, we obtain for the impedance

of the cavity:

p Ks

Xa = = . (88)

U jωV

We see that a cavity at low frequency presents a capacitive reactance: The

air acts as a spring that is loaded by the incident pressure.

Finally, if we connect the small tube of (85) to the opening of the cavity of

(88), we obtain a configuration as in Fig. 28. The total acoustic impedance will

be the sum of the two. Using Ks = ρ0 c20 :

ρ0 c20

Za = jωρ0 lef /S + . (89)

jωV

It is easily verified that Za goes to zero when the frequency approaches

s

S

ω0 = c0 . (90)

V lef

This is the resonant frequency of the system of Fig. 28. Such a system is

called a Helmholtz resonator. It is the acoustic equivalent of the mechanical

54

Figure 28: Short open tube of length l and cross section S connected to a small

cavity of arbitrary shape and volume V .

resonator in Fig. 27. In both cases, the slightest driving force at the resonant

frequency will result in an arbitrarily large oscillation (in reality the impedance

will of course never be exactly zero because there will always be resistive losses).

Notice the similarity of the Helmhotlz resonator to an empty bottle. If we blow

across the neck of the empty bottle, our breath will cause random turbulences,

a small part of which is enough to excite the resonance and produce the typical

sound. Helmholtz resonators in various appearances are an important concept

in acoustics.

In the previous section we saw that short segments of pipes act as an acoustic

mass if they are open-ended and as an acoustic spring if they are closed. When

they are neither open or closed but matched, that is, terminated without any

reflections, then the impedance in a cross-section of the pipe equals that of a

propagating plane wave:

Z0

Za = . (91)

S

Observe that the impedance changes when the cross-section area S changes.

Now, as an example, consider the sequence of segments depicted in Fig. 29.

The presence of the wide segment in the middle can be modeled a continuous

pipe of cross-section S in parallel with a closed cavity of volume V = L (S1 − S).

The reflection coefficient of this circuit is readily calculated as

1

R= 2c0 S

(92)

1+ jωV

2

which represents a low-pass filter with a cutoff-frequency (when |R| = 1/2)

of ωc = 2c0 S/V . A good reference with more examples of elementary acoustic

circuits is [17]. The analogy with electrical circuit theory is evident in the above.

As is the case with electrical circuits, when the length of the pipe segments is

longer than a fraction of the wavelength, the lumped element model ceases to be

valid and we have to allow for plane waves travelling up and down the segment.

55

Figure 29: Acoustic Low-pass filter made of a pipe of cross-section S with an

enlarged segment of length L and cross-section S1 .

References

[1] https://www.isover.co.za/acoustic

[2] https://ebom.com/2018/05/arduino-sound-level-meter/

[3] https://www.castlegroup.co.uk

[4] http://www.industrial-electronics.com/transformers_6c.html

[5] https://en.wikipedia.org/wiki/Noise_curve

[6] https://www.dpamicrophones.com/mic-university/facts-about-speech-

intelligibility

[7] http://home.cc.umanitoba.ca/~robh/archives/arc0705.html

[8] http://en.wikipedia.org/wiki/File:Spectrogram_of_violin.png

[9] https://www.researchgate.net/publication/

292150038_Fundamentals_of_Acoustics_and_Noise_Control

[10] J. Acoust. Soc. Am., Vol 29, p. 743 (1957)

[11] J. Acoust. Soc. Am., vol.40, pp. 148-159 (1966)

[12] http://dev.physicslab.org

[13] H. Kuttruff, Room Acoustics, 5th Ed. (2009)

[14] http://zoneplate.lbl.gov/theory

[15] http://acoustics.group.shef.ac.uk/asaproject.php

[16] https://rosap.ntl.bts.gov/view/dot/30259/dot_30259_DS1.pdf

[17] Fundamentals of Acoustics, 4th Ed. L. E. Kinsler ea.

56